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libavcodec
flacdsp_template.c
Go to the documentation of this file.
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/*
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* Copyright (c) 2012 Mans Rullgard <mans@mansr.com>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <stdint.h>
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#include "
libavutil/avutil.h
"
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#undef FUNC
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#undef FSUF
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#undef sample
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#undef sample_type
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#undef OUT
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#undef S
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#if SAMPLE_SIZE == 32
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# define sample_type int32_t
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#else
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# define sample_type int16_t
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#endif
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#if PLANAR
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# define FSUF AV_JOIN(SAMPLE_SIZE, p)
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# define sample sample_type *
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# define OUT(n) n
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# define S(s, c, i) (s[c][i])
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#else
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# define FSUF SAMPLE_SIZE
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# define sample sample_type
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# define OUT(n) n[0]
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# define S(s, c, i) (*s++)
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#endif
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#define FUNC(n) AV_JOIN(n ## _, FSUF)
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static
void
FUNC
(
flac_decorrelate_indep_c
)(
uint8_t
**
out
,
int32_t
**
in
,
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int
channels,
int
len
,
int
shift
)
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{
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sample
*samples = (
sample
*)
OUT
(
out
);
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int
i, j;
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for
(j = 0; j <
len
; j++)
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for
(i = 0; i < channels; i++)
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S
(samples, i, j) =
in
[i][j] <<
shift
;
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}
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static
void
FUNC
(
flac_decorrelate_ls_c
)(
uint8_t
**
out
,
int32_t
**
in
,
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int
channels,
int
len
,
int
shift
)
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{
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sample
*samples = (
sample
*)
OUT
(
out
);
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int
i;
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for
(i = 0; i <
len
; i++) {
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int
a
=
in
[0][i];
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int
b
=
in
[1][i];
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S
(samples, 0, i) = a <<
shift
;
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S
(samples, 1, i) = (a -
b
) << shift;
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}
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}
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static
void
FUNC
(
flac_decorrelate_rs_c
)(
uint8_t
**
out
,
int32_t
**
in
,
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int
channels,
int
len
,
int
shift
)
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{
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sample
*samples = (
sample
*)
OUT
(
out
);
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int
i;
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for
(i = 0; i <
len
; i++) {
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int
a
=
in
[0][i];
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int
b
=
in
[1][i];
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S
(samples, 0, i) = (a +
b
) <<
shift
;
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S
(samples, 1, i) = b <<
shift
;
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}
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}
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static
void
FUNC
(
flac_decorrelate_ms_c
)(
uint8_t
**
out
,
int32_t
**
in
,
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int
channels,
int
len
,
int
shift
)
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{
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sample
*samples = (
sample
*)
OUT
(
out
);
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int
i;
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for
(i = 0; i <
len
; i++) {
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int
a
=
in
[0][i];
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int
b
=
in
[1][i];
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a -= b >> 1;
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S
(samples, 0, i) = (a +
b
) <<
shift
;
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S
(samples, 1, i) = a <<
shift
;
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}
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}
Generated on Sun Mar 23 2014 23:49:54 for FFmpeg by
1.8.2